similar to: What can I do with Jabber?

Hello I've been googling for this message, but can't find why Asterisk sends a warning. The configuration files look similar to http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf.sample It's a TDM card with just one FXO module on it, and I connected an RJ11 cable to it from the wall plug: # cat /etc/zaptel.conf fxsks=1 loadzone=fr defaultzone=fr # ztcfg -vv

Hi all, I was wondering if someone could explain the relationship of Asterisk, SER & Jabber (XMPP) to me. I understand that there are facilities within Asterisk to use jabber to notify of incoming calls via XMPP clients, however I'm trying to work out exactly where the SER server would sit in all of this and what it's actual role would be. This might seem like a silly

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

Hello I'm curious about what can be done when using Jabber with Asterisk. What are good examples of this combination? Thanks.

Hello A user seems unable to type DTMF in our Asterisk IVR menu. Can this be due to their phone or PBX that disables DTMFs when a user is off-hook? Thank you.

Hi all, All the examples I've come across seem to suggest configuring jabber.conf/jingle.conf/gtalk.conf for a real Google account. What about the scenario where the Asterisk server should connect to an account on a private Jabber server and using Jingle (voice calling over Jabber)? e.g. for the domain widgets.com: - there is a copy of ejabberd running on the same box as Asterisk, and

Hello About Record(), ATFT 2nd Edition says that "if the filename contains %d, these characters will be replaced with a number incremented by one each time the file is recorded." Problem is, the documentation doesn't explain how to refer to this filename later in the dialplan :-/ In this particular example, I want to move the file to the web server's /htdocs so users can

I have installed gnutls and gnutls-devel from RedHat repositories [root at asterisk asterisk]# yum install gnutls gnutls-devel I have installed iksemel with gnutls support : [root at asterisk asterisk]# cd /usr/src/iksemel-1.3/ [root at asterisk asterisk]# ./configure --with-gnutls --prefix=/usr [root at asterisk asterisk]# make [root at asterisk asterisk]# make check [root at asterisk

Hi Ulrich, <p>you distroyed my evening today ;-). Im only sitting here and follow the videoconferencing thread ;-)). > due to the ongoing discussion on both lists, i simply respond to both > lists. it's hard crossposting, but it's for both roups relevant (i > think). Yes, this was right. There is a great discussion and perhaps we got some points more yet. > +After

Hello When using LookupCIDName, Asterisk 1.4 says that it's deprecated, and we should use "${DB(cidname/${CALLERID(num)})}" instead, but I don't know how to use it: ;DEPRECATED exten => s,1,LookupCIDName ;ERROR exten => s,1,${DB(cidname/${CALLERID(num)})} I guess I should use this as a parameter to a function, but which one? Thank you.

Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you.

Hi All, In order to enable chat functionality with gmail in my rails application i used "Jabber" and "xmpp4r/simple" and finished the chat functionality well and fine. Now i was asked to enable the chat functionality with other services like "MSN","Yahoo","AOL" etc which requires a third party server to act as mediator between the service(like

I've included my jabber.conf below. I'm betting the following errors: [Jul 18 21:05:22] ERROR[28166]: res_jabber.c:609 aji_act_hook: JABBER: Node Error [Jul 18 21:05:22] WARNING[28166]: res_jabber.c:1537 aji_recv_loop: JABBER: Got hook event. jabber test [Jul 18 21:04:16] WARNING[32691]: res_jabber.c:1421 ast_aji_send: JABBER: Not connected can't send User: bferrell at gtalk.com

Speaking of video conferencing in conjunction with Speex, we are currently beta testing a solution we developed that offers multi-point, multi-party video and audio using the Speex engine for voice. http://www.wiredred.com/downloads/ecsetup.exe The fair and good audio settings are Speex narrowband, high quality is HawkVoice and toll quality is PCM. -----Original Message----- From:

Hi all, Do some one experiencing running jabber applications (jabberstatus...) in asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I got such result. IBM*CLI> help jabber No such command 'jabber'. IBM*CLI> help jabberstatus No such command 'jabberstatus'. Any one can help me on this, or may be I miss out somethings that cause jabber applications

I want to use JabberSend in my dialplan, but I saw that my Asterisk does not support Jabber. Also I have nowhere a module res_jabber.so... So I thought I'd rebuild my Asterisk. In menuselect I saw that res_jabber was dependent of 'iksemel' and 'gnutls'. In my yum repositories I can find a gnutls.i386, but what is this iksemel-beast ??? There is info to find via google on

Hi all, Do some one experiencing running jabber applications (jabberstatus...) in asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I got such result. IBM*CLI> help jabber No such command 'jabber'. IBM*CLI> help jabberstatus No such command 'jabberstatus'. Any one can help me on this, or may be I miss out somethings that cause jabber applications

Hi Carsten, due to the ongoing discussion on both lists, i simply respond to both lists. it's hard crossposting, but it's for both roups relevant (i think). <p>+After having thought about control structures, it makes sense to me to do the extra work and merge this creamed cake into a jabber server component. Otherwise a control channel to the server component would have to be

asterisk-1.8.13.0 iksemel-1.4 I have a client who setup a gvoice account using their domain in the login name: username=client at theirdomain@gmail.com This appears to have caused a problem with authentication. I've tried escaping the @ and quoting the login string, etc. but it simply won't authenticate. I don't believe my configuration is bad as the same server /

Greetings, I hope this message is not too far off-topic. I'm working with the Helix-Jabber project to create a standard for voice and video with Jabber IM (http://jabber.org). I've received numerous recommendations about Speex, and so for our standard voice codec, I'm wondering if it would be wise to choose Speex instead of GSM. The codecs we choose must be accessible to as